Vonage "hard" lines only run through an FXO port. There's no feasible way to get the SIP credentials in order to terminate directly into *. You can terminate a "soft" line directly into asterisk, both for incoming and outgoing calls -- for configuration examples, simply search the list archives (use google and add "site:lists.digium.com" to your search-term).
> -----Original Message----- > From: Angus Berry [mailto:[EMAIL PROTECTED] > Sent: Monday, November 29, 2004 12:27 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Vonage integration... Hardware or > Softphone typeacct. > > > Hi All, > > I've got an * PBX up with couple of stations and now I'd like > to integrate my Vonage service for outgoing PSTN calls. Is > this possible if I have an account with them that uses their > hardware box (ATA186) or do I need a 'softphone' account? > > thanks... > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
